The present invention relates generally to the field of processing telecommunications signals. More particularly, the invention provides a method and apparatus for voice transcoding from a CELP based voice compression codec to a hybrid based voice compression codec (i.e. a codec that uses both CELP and non-CELP parameters). Merely by way of example, the invention has been applied to transcoding from the GSM-AMR codec to the internet Low Bitrate Codec (iLBC), but it would be recognized that the invention may also include other applications.
Modern communication systems rarely transmit uncompressed signals. Instead, signals are compressed to allow efficient utilization of spectrum resources. Compression of signals is generally performed by removing statistical and perceptual redundancy in the signal. In the process of compression, a block (known as a frame) of uncompressed samples is represented by a set (also known as a frame) of compression parameters. The compression parameters are subsequently quantized. The quantization indices for the compression parameters are organized into a bitstream. In the decompression process, the quantized compression parameters are extracted from the bitstream and used to construct a signal that replicates the original and may or may not be exactly the same. Typically, compression systems aim to produce perceptually similar signals to the original but in some cases exact replicas are also produced.
A number of standardized compression systems, which will from this point on be referred to as codecs, are based on the Code Excited Linear Prediction (CELP) algorithm (for example, the ITU's G.723.1 and the GSM's AMR codecs). CELP based codecs are popular for speech signal compression in mobile networks. CELP based codecs represent a speech signal by a linear prediction filter and an excitation signal. The excitation signal is vector quantized with a codebook that contains an adaptive section (referred to as the adaptive codebook, in which the code words are constructed from past quantized excitation signal samples) and a fixed or innovation section (where the code words are extracted from a static codebook).
Different networks follow different formats in compressing signals (i.e., different terminals on the same network may also use different formats). Recently, the internet Low Bit-rate Codec (iLBC),has been introduced for voice over internet protocol (VoIP) applications. The main feature that makes iLBC suitable for VoIP application is its graceful performance degradation in the presence of packet loss, which is typical in Internet Protocol (IP) networks. Packet loss tolerance is achieved by quantizing the excitation signal of each frame independently of other frames.
In order to ensure that different terminals using different audio (of which speech is a subset) codecs can communicate, converting bitstreams of different formats is generally necessary. A straightforward way of carrying out a bitstream conversion task is by cascading a source bitstream decoder and a destination bitstream encoder in sequence. This is known as the tandem solution. Although the tandem solution is conceptually simple, actual implementation generally requires extensive computations and a tandem solution does not make effective use of the parameters used in the already encoded incoming bitstream. Thus, there is a need in the art for improved methods and systems for transcoding CELP based voice compression codec to a hybrid based voice compression codec in a more efficient manner.